On 08/16/2013 12:29 PM, Roberto Fichera wrote: > On 08/14/2013 09:58 PM, Vitaliy Aleksandrov wrote: >> On 08/14/2013 07:32 PM, Roberto Fichera wrote: >>> On 08/14/2013 04:36 PM, Vitaliy Aleksandrov wrote: >>>> If you won't be able to disable SIP ALG on your router you can fill >>>> $avp(received) manually before calling save(): >>>> $avp(received) = "sip:" + $si + ":" + $sp + ";transport=" + $proto; >>>> >>>> In this case all user location records will have the "received" attribut >>>> even if a UA isn't behind NAT, but I don't >>>> see any problems with that. >>> This one looks working, but the callee doesn't answer correctly because the >>> TCP isn't correct: >>> >>> Contact:: >>> <sip:528@94.94.X.X:1380;transport=TCP;ob>;q=;expires=294;flags=0x0;cflags=0x0;state=0;socket=<tcp:178.79.X.X:5060>;methods=0x1FDF;received=<sip:94.94.X.X:37030;transport=tcp>;user_agent=<PJSUA >>> >>> v2.1-svn Linux-3.10.5.201/x86_64/glibc-2.17>;reg-id=0 >>> >>> The contact uses a port which isn't translated inside by the router, the >>> received field shows the right one. >>> Should I change the Contact header instead? In case, how can I do that? >> I didn't understand what "callee doesn't answer correctly" means. >> Callee doesn't know what's in the received field of its registration. > Sorry! My typo! I meant why kamilio wasn't reusing the TCP port specified in > the REGISTER even for the INVITE. > I mean, kamailio now knows the received field so I'm expecting it routes the > requests for the called UA through the > address:port specified in this field. > >> The only problem I see is that your router changes ip in the contact field >> of REGISTER requests and then kamailio puts >> this value >> (new_ip:old_port) to INVITEs destined to UAs behind your NAT router. > Indeed, my testing router is affected by this ALG "problem"! I guess it's > better to disable it and complete simple TCP tests > then move to TLS. > >> It's not likely, but maybe pjsip doesn't like INVITEs with RURI which >> differs from what it put to the contact during >> registration. >> IIRC pjsua prints lots of debugging information. So you can check if pjsua >> shows anything when INVITE comes in its log >> and also attach a trace of such a call. > pjsua doesn't report any problem so far, so seems accepting the INVITE. > >> P.S. IMHO the best way to pass sip through such routers is sip over TLS. > Yep! I'll do that.
Just to say that everything works pretty fine once moved to TLS! > >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
